1 files changed, 18 insertions, 10 deletions

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index ba78d4e988..b64010ff44 100644
--- a/libavformat/audiointerleave.c
+++ b/libavformat/audiointerleave.c
@@ -3,20 +3,20 @@
  *
  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -34,7 +34,7 @@ void ff_audio_interleave_close(AVFormatContext *s)
         AudioInterleaveContext *aic = st->priv_data;
 
         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
-            av_fifo_free(aic->fifo);
+            av_fifo_freep(&aic->fifo);
     }
 }
 
@@ -45,8 +45,12 @@ int ff_audio_interleave_init(AVFormatContext *s,
     int i;
 
     if (!samples_per_frame)
-        return -1;
+        return AVERROR(EINVAL);
 
+    if (!time_base.num) {
+        av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
+        return AVERROR(EINVAL);
+    }
     for (i = 0; i < s->nb_streams; i++) {
         AVStream *st = s->streams[i];
         AudioInterleaveContext *aic = st->priv_data;
@@ -56,14 +60,15 @@ int ff_audio_interleave_init(AVFormatContext *s,
                                 av_get_bits_per_sample(st->codec->codec_id)) / 8;
             if (!aic->sample_size) {
                 av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
-                return -1;
+                return AVERROR(EINVAL);
             }
             aic->samples_per_frame = samples_per_frame;
             aic->samples = aic->samples_per_frame;
             aic->time_base = time_base;
 
             aic->fifo_size = 100* *aic->samples;
-            aic->fifo= av_fifo_alloc(100 * *aic->samples);
+            if (!(aic->fifo= av_fifo_alloc_array(100, *aic->samples)))
+                return AVERROR(ENOMEM);
         }
     }
 
@@ -110,7 +115,7 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt
             unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
             if (new_size > aic->fifo_size) {
                 if (av_fifo_realloc2(aic->fifo, new_size) < 0)
-                    return -1;
+                    return AVERROR(ENOMEM);
                 aic->fifo_size = new_size;
             }
             av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
@@ -128,9 +133,12 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt
         AVStream *st = s->streams[i];
         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             AVPacket new_pkt;
-            while (interleave_new_audio_packet(s, &new_pkt, i, flush))
+            while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
                 if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
                     return ret;
+            }
+            if (ret < 0)
+                return ret;
         }
     }